FFmpeg
af_flanger.c
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1 /*
2  * Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net>
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "libavutil/avstring.h"
22 #include "libavutil/opt.h"
23 #include "libavutil/samplefmt.h"
24 #include "avfilter.h"
25 #include "audio.h"
26 #include "internal.h"
27 #include "generate_wave_table.h"
28 
29 #define INTERPOLATION_LINEAR 0
30 #define INTERPOLATION_QUADRATIC 1
31 
32 typedef struct FlangerContext {
33  const AVClass *class;
34  double delay_min;
35  double delay_depth;
36  double feedback_gain;
37  double delay_gain;
38  double speed;
40  double channel_phase;
42  double in_gain;
46  double *delay_last;
47  float *lfo;
49  int lfo_pos;
51 
52 #define OFFSET(x) offsetof(FlangerContext, x)
53 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
54 
55 static const AVOption flanger_options[] = {
56  { "delay", "base delay in milliseconds", OFFSET(delay_min), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
57  { "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
58  { "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
59  { "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
60  { "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
61  { "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" },
62  { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
63  { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
64  { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
65  { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
66  { "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
67  { "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" },
68  { "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, "itype" },
69  { "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" },
70  { NULL }
71 };
72 
73 AVFILTER_DEFINE_CLASS(flanger);
74 
75 static int init(AVFilterContext *ctx)
76 {
77  FlangerContext *s = ctx->priv;
78 
79  s->feedback_gain /= 100;
80  s->delay_gain /= 100;
81  s->channel_phase /= 100;
82  s->delay_min /= 1000;
83  s->delay_depth /= 1000;
84  s->in_gain = 1 / (1 + s->delay_gain);
85  s->delay_gain /= 1 + s->delay_gain;
86  s->delay_gain *= 1 - fabs(s->feedback_gain);
87 
88  return 0;
89 }
90 
92 {
95  static const enum AVSampleFormat sample_fmts[] = {
97  };
98  int ret;
99 
100  layouts = ff_all_channel_counts();
101  if (!layouts)
102  return AVERROR(ENOMEM);
103  ret = ff_set_common_channel_layouts(ctx, layouts);
104  if (ret < 0)
105  return ret;
106 
107  formats = ff_make_format_list(sample_fmts);
108  if (!formats)
109  return AVERROR(ENOMEM);
110  ret = ff_set_common_formats(ctx, formats);
111  if (ret < 0)
112  return ret;
113 
114  formats = ff_all_samplerates();
115  if (!formats)
116  return AVERROR(ENOMEM);
117  return ff_set_common_samplerates(ctx, formats);
118 }
119 
121 {
122  AVFilterContext *ctx = inlink->dst;
123  FlangerContext *s = ctx->priv;
124 
125  s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
126  s->lfo_length = inlink->sample_rate / s->speed;
127  s->delay_last = av_calloc(inlink->channels, sizeof(*s->delay_last));
128  s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo));
129  if (!s->lfo || !s->delay_last)
130  return AVERROR(ENOMEM);
131 
133  rint(s->delay_min * inlink->sample_rate),
134  s->max_samples - 2., 3 * M_PI_2);
135 
137  inlink->channels, s->max_samples,
138  inlink->format, 0);
139 }
140 
142 {
143  AVFilterContext *ctx = inlink->dst;
144  FlangerContext *s = ctx->priv;
145  AVFrame *out_frame;
146  int chan, i;
147 
148  if (av_frame_is_writable(frame)) {
149  out_frame = frame;
150  } else {
151  out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
152  if (!out_frame) {
153  av_frame_free(&frame);
154  return AVERROR(ENOMEM);
155  }
156  av_frame_copy_props(out_frame, frame);
157  }
158 
159  for (i = 0; i < frame->nb_samples; i++) {
160 
161  s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;
162 
163  for (chan = 0; chan < inlink->channels; chan++) {
164  double *src = (double *)frame->extended_data[chan];
165  double *dst = (double *)out_frame->extended_data[chan];
166  double delayed_0, delayed_1;
167  double delayed;
168  double in, out;
169  int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
170  double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
171  int int_delay = (int)delay;
172  double frac_delay = modf(delay, &delay);
173  double *delay_buffer = (double *)s->delay_buffer[chan];
174 
175  in = src[i];
176  delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
177  s->feedback_gain;
178  delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
179  delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
180 
182  delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
183  } else {
184  double a, b;
185  double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
186  delayed_2 -= delayed_0;
187  delayed_1 -= delayed_0;
188  a = delayed_2 * .5 - delayed_1;
189  b = delayed_1 * 2 - delayed_2 *.5;
190  delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
191  }
192 
193  s->delay_last[chan] = delayed;
194  out = in * s->in_gain + delayed * s->delay_gain;
195  dst[i] = out;
196  }
197  s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
198  }
199 
200  if (frame != out_frame)
201  av_frame_free(&frame);
202 
203  return ff_filter_frame(ctx->outputs[0], out_frame);
204 }
205 
207 {
208  FlangerContext *s = ctx->priv;
209 
210  av_freep(&s->lfo);
211  av_freep(&s->delay_last);
212 
213  if (s->delay_buffer)
214  av_freep(&s->delay_buffer[0]);
215  av_freep(&s->delay_buffer);
216 }
217 
218 static const AVFilterPad flanger_inputs[] = {
219  {
220  .name = "default",
221  .type = AVMEDIA_TYPE_AUDIO,
222  .config_props = config_input,
223  .filter_frame = filter_frame,
224  },
225  { NULL }
226 };
227 
228 static const AVFilterPad flanger_outputs[] = {
229  {
230  .name = "default",
231  .type = AVMEDIA_TYPE_AUDIO,
232  },
233  { NULL }
234 };
235 
237  .name = "flanger",
238  .description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
239  .query_formats = query_formats,
240  .priv_size = sizeof(FlangerContext),
241  .priv_class = &flanger_class,
242  .init = init,
243  .uninit = uninit,
244  .inputs = flanger_inputs,
245  .outputs = flanger_outputs,
246 };
#define NULL
Definition: coverity.c:32
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:549
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
AVOption.
Definition: opt.h:246
double * delay_last
Definition: af_flanger.c:46
double feedback_gain
Definition: af_flanger.c:36
Main libavfilter public API header.
float * lfo
Definition: af_flanger.c:47
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
Definition: af_flanger.c:141
double, planar
Definition: samplefmt.h:70
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:36
uint8_t ** delay_buffer
Definition: af_flanger.c:44
AVFilter ff_af_flanger
Definition: af_flanger.c:236
#define src
Definition: vp8dsp.c:254
int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Allocate a data pointers array, samples buffer for nb_samples samples, and fill data pointers and lin...
Definition: samplefmt.c:198
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
const char * name
Pad name.
Definition: internal.h:60
static int query_formats(AVFilterContext *ctx)
Definition: af_flanger.c:91
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
#define A
Definition: af_flanger.c:53
uint8_t
#define av_cold
Definition: attributes.h:82
AVOptions.
A filter pad used for either input or output.
Definition: internal.h:54
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
double channel_phase
Definition: af_flanger.c:40
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:568
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:86
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
static const AVFilterPad flanger_outputs[]
Definition: af_flanger.c:228
double delay_depth
Definition: af_flanger.c:35
#define b
Definition: input.c:41
#define M_PI_2
Definition: mathematics.h:55
double in_gain
Definition: af_flanger.c:42
AVFormatContext * ctx
Definition: movenc.c:48
AVFILTER_DEFINE_CLASS(flanger)
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
#define s(width, name)
Definition: cbs_vp9.c:257
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
void ff_generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt, void *table, int table_size, double min, double max, double phase)
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_flanger.c:206
#define INTERPOLATION_LINEAR
Definition: af_flanger.c:29
#define rint
Definition: tablegen.h:41
A list of supported channel layouts.
Definition: formats.h:85
if(ret)
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
double speed
Definition: af_flanger.c:38
static const AVOption flanger_options[]
Definition: af_flanger.c:55
#define INTERPOLATION_QUADRATIC
Definition: af_flanger.c:30
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
#define OFFSET(x)
Definition: af_flanger.c:52
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);return NULL;}return ac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
double delay_gain
Definition: af_flanger.c:37
static const AVFilterPad flanger_inputs[]
Definition: af_flanger.c:218
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
static int config_input(AVFilterLink *inlink)
Definition: af_flanger.c:120
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
double delay_min
Definition: af_flanger.c:34
static int init(AVFilterContext *ctx)
Definition: af_flanger.c:75
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:338
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
FILE * out
Definition: movenc.c:54
#define av_freep(p)
formats
Definition: signature.h:48
internal API functions
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654